Secure communication system

ABSTRACT

A communication system including: a first secure communication terminal for providing analog voiceband data; a first processing circuit connected to receive the secure data from the first secure communication terminal and for converting the received data into secure baseband data, the first processing circuit transmitting the baseband data; a second processing circuit connected to receive--the transmitted baseband data from the first processing circuit, and for converting the received baseband data into analog voiceband data; and a second secure communication terminal for receiving the analog voiceband data from the second processing circuit. The communication system is especially applicable for handling secure data transmitted by a STU-III.

This is a continuation of application Ser. No. 08/324,480 filed Oct. 17,1994, now U.S. Pat. No. 5,784,414, which is a continuation-in-part ofapplication Ser. No. 08/065,017 filed May 24, 1993 now U.S Pat. No.5,404,394.

FIELD OF THE INVENTION

The present invention relates in general to a system for supporting thedirect interfacing of secure communication services, and in particular,to a system for supporting the direct interfacing of the U.S.Government's Secure Terminal Unit-III (STU-III) over low-rate digitalmobile systems.

BACKGROUND OF THE INVENTION

The rapid increase in the popularity and use of cellular and satellitecommunications systems has highlighted the impeding congestion of mobilecommunications spectrum. Concurrently, rapid advances in digital signalprocessing technology have made possible the transmission of voicesignals at ever decreasing encoding rates. This latter development ispermitting the transmission of speech to be realized over much narrowerchannel bandwidths, thus promising to increase the system's subscribercapacity proportionally to the potential bandwidth requirementsreduction (e.g., five-fold). These two developments have permittedmobile networks to be converted from analog-based systems todigital-based systems.

Unfortunately, the digitization of new mobile networks has made thetransparent support of non-voice traffic impossible without thedevelopment of specialized interworking functions, which are tailored tospecific end-user applications.

SUMMARY OF THE INVENTION

It is a general object of the present invention to provide a system forsupporting a secure system over a digital system.

It is another object of the invention to provide a system for supportingthe Secure Terminal Unit Version-III (STU-III) over low-rate digitalsystems including mobile systems.

These and other objects are realized by the present secure communicationsystem.

In the above-mentioned U.S. application Ser. No. 08/065,017, a securecommunication system is described for solving the problem of enablingcustomers of STU-III terminals operating over the analog public switchedtelephone network (PSTN) the ability to communicate over, for example,the Inmarsat-M digital satellite network. In particular, the securecommunication system defines an inter-working specification whichprovides interoperability of analog STU-III equipment for use, forexample, over the Inmarsat-M network.

As with Inmarsat-M channels, the Inmarsat-B is an all-digital system.The advantages of providing all-digital services is that satelliteservice providers are able to offer additional capacity to accommodatemore subscribers and higher traffic levels than with analog satellitechannels. This means that the digital service offered through Inmarsat-B(as with Inmarsat-M) terminals will be more efficient in their use ofspectrum, providing better quality services in a more cost-effectivemanner. The primary difference between Inmarsat-M and Inmarsat-B systemsis that the Inmarsat-M is a smaller, less cost effective terminal thanthe Inmarsat-B. Inmarsat-M services are designed for smaller craft andmobile vehicles which also have a need to communicate with shore-sidepoints but do not have the means or need for larger Inmarsat-Bterminals.

Unlike the Inmarsat-M system, Inmarsat-B service is capable of directlypassing the required narrow-band tones through the voice coder. Theexchange of narrow-band tones between STU-III terminals is essential forsecure call establishment. In Inmarsat-M, the voice codec is able torepeatedly detect whether the incoming signal is either voice or anarrow-band tone. When a narrow-band tone is detected, a special bitsequence is transmitted which corresponds to an invalid voice signal. Atthe receiving earth-station, when an invalid voice is detected, thedecoder regenerates the narrow-band tone as designated by the specialbit sequence. However, for the Inmarsat-B system, since the channelvoice coder is capable of accurately passing narrow-band signalsdirectly (e.g., P1800 and 2100 Hz), it is unnecessary to modify thevoice coder to enable the passage of such tones.

Similar to the Inmarsat-M system, Inmarsat-B employs the same type ofchannel transport mechanism for both voice and secure traffic. However,since the Inmarsat-B system channel format is significantly differentfrom the Inmarsat-M system, a different channel structure has beendefined for error coding and rate adaptation. Likewise, since the voicefield channel structure and error characteristics for the Inmarsat-Bsystem are notably different, a different error coding scheme has beendesigned for the two currently supported secure data rates.

Similar to the Inmarsat-M system, the Inmarsat-B secure protocol usessecure protocol packets to convey information regarding the initiationand termination of signal activity. However, in the Inmarsat-Barchitecture, all secure protocol packets are transmitted in triplicatefor reliability.

DETAILED DESCRIPTION OF THE INVENTION

According to the present invention, the interworking functionality andarchitecture are defined for supporting the direct interfacing of theU.S. Government's Secure Terminal Unit-III (STU-III) over low-ratedigital mobile systems. Although this functionality is presented from aparticular system perspective, it is applicable to other low-ratedigital mobile systems.

The functionality is described by means of a Secure Interface Unit (SIU)architecture and a Secure Communications (SC) protocol.

The primary function of the SIU architecture is the demodulation ofsecure traffic in the secure terminal-to-satellite direction and there-modulation (or regeneration) of signals in the mobile- orfixed-station-to-secure terminal direction. In addition to the securemodems, the SIU incorporates a rate adaptation and protocolpacketization mechanism, a channel encoding process, an elastic buffer,and an echo canceller.

This SC protocol is based on a packetized structure that interfaces withthe voice channel to provide secure-to-clear and clear-to-securecommunication in a seamless manner. In addition, this functionalityprovides for the controlled and synchronized call establishment to occuron an end-to-end basis, while it provisions for double satellite hopconditions. The SC protocol enhances the voice channel coding by usingan error code to provide high-quality end-user communications even underlow link-margin conditions.

The SIU is located at, for example, mobile and fixed stations, where itinterfaces with the end-user secure terminals via 2-wire connections.The SIU is able to support a number of STU-III features including fulland half-duplex operation at 2400 bit/s (interoperable) and 4800 bit/s(alternate) autosecure on receive and plan-text inhibit; and synchronoussecure data.

The packetization approach, error-coding scheme, channel structure andhandling of the call during its establishment and clearing are some ofthe features of the communication system according to the presentinvention.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a general block diagram showing a network configurationaccording to the present invention.

FIG. 2 is a state transition diagram for explaining the operation of aSTU-III Terminal.

FIG. 3 is a block diagram of a Mobile Earth Station for Voice andNon-Voice Services according to an embodiment of the invention.

FIG. 4 is a block diagram of a Secure Interface Unit (SIU) according toan embodiment of the invention.

FIG. 5(a) is a circuit diagram of a Rate 3/8 Augmented ConvolutionalCode Generator of the invention.

FIG. 5(b) is a diagram for illustrating the composition of voice andassociated fields during 2400 bit/s secure communication.

FIG. 6(a) is a circuit diagram of a Rate 3/4 Punctured ConvolutionalCode Generator according to embodiment of the invention.

FIG. 6(b) is a diagram for illustrating the composition of voice andassociated fields during 4800 bit/s secure communication.

FIG. 7 is a state diagram for Voice-Non-Voice Call Discriminationaccording to the invention.

FIG. 8 is a diagram for illustrating secure Voice Call Establishmentover the PSTN (full-duplex) according to the invention.

FIG. 9 is a diagram for illustrating Secure Voice Call Establishmentover the PSTN (half-duplex) according to the invention.

FIG. 10 is a diagram for illustrating Facsimile Call Establishment overthe PSTN according to the invention.

FIG. 11 is a diagram for illustrating Secure Call Establishment(full-duplex) according to the invention.

FIG. 12 is a diagram for illustrating Secure (full-duplex) CallEstablishment Process according to the invention.

FIG. 13 is a diagram for illustrating the Secure Call Establishment(half-duplex) according to the invention.

FIG. 14 is a diagram for illustrating Secure (half-duplex) CallEstablish Process according to the invention.

FIGS. 15(a) and 15(b) show a table for explaining packet types anddesignations according to the invention.

FIG. 16(A) is a diagram for explaining the structure of Secure CallChannel Establishment Packets according to an embodiment of theinvention.

FIG. 16(b) is a diagram for explaining the structure of Secure CallChannel Establishment Packets according to another embodiment of theinvention.

FIG. 17(A) is a diagram for explaining the structure of secure ProtocolControl Packets according to an embodiment of the invention.

FIG. 17(B) is a diagram for explaining the structure of secure ProtocolControl Packets according to another embodiment of the invention.

FIG. 17(C) is a diagram for explaining the structure of secure ProtocolControl Information field according to an embodiment of the invention.

FIGS. 18(A) and 18(B) are diagrams for explaining Secure TransmissionProtocol Processing of P1800 Signal according to embodiments of theinvention.

FIGS. 19(a)-19(c) are tables for explaining the processing by anInitiating Secure Interface Unit according to the invention.

FIG. 20 is a diagram for explaining the Secure Transmission ProtocolProcessing of an MSG A Signal according to the invention.

FIG. 21 is a diagram for explaining the Secure Transmission ProtocolProcessing of an MSG B Signal according to the invention.

FIG. 22 is a block diagram for explaining Modem Training Sequenceaccording to the invention.

FIGS. 23(a)-23(d) are diagrams for explaining the Generation of aTraining Sequence according to the invention.

FIG. 24 is a table for explaining coding according to a feature of thepresent invention.

FIG. 25 is a diagram for explaining Secure Transmission ProtocolProcessing of a Responder's Signal according to the invention.

FIG. 26 is a diagram for explaining the Secure Transmission ProtocolProcessing of an Initiator's Signal according to the invention.

FIG. 27 is a diagram for explaining Modem Training according to ahalf-duplex mode of the invention.

FIG. 28 is a diagram for explaining Transition Times for Abort SecureCall Conditions according to the invention.

FIG. 29 is a state diagram for explaining Call Interruption Conditionsaccording to the invention.

FIG. 30 is a circuit diagram for explaining the Echo ControlConfiguration according to the invention.

FIGS. 31(a)-31(c) are tables for explaining the format of Data Vectorsaccording to the invention.

FIG. 32 is a table for describing the Tone Index according to theinvention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 shows a general network configuration according to an embodimentof the present invention. As shown in FIG. 1, the network includes aLand Earth Station 2, and a Mobile Earth Station 4 which is connected tothe Land Earth Station 2 via a satellite communication link 6. As shownin FIG. 1, a Public Switched Telephone Network (PSTN) 8 connects POTs(Plain old Telephones) and STU-III terminals to the Land Earth Station2. On the other hand, POTs and STU-III terminals are directly connectedto the Mobile Earth Station 4 which may be for example, a ship.

As shown in FIG. 1, the Land Earth Station 2 includes a FacilitiesAssignment Processor 9, a Secure Interface Unit (SIU) 10, a DigitalVoice Codec 12, a Mux & Channel Unit 13 and a Station Control unit 14.The Facilities and Assignment Processor 9 is coupled between the PSTN 8,and the SIU 10 and the Digital Voice Codec 12, and functions to assignthe various elements of the Land Earth Station 2 to the type of phoneservice requested. For example, the Facilities Assignment Processor 9would assign the Digital Voice Codec when POT service is requested, andwould assign the SIU 10 when a secure phone service is requested. TheMux & channel Unit 13 controls the transfer of data to and from thesatellite link 6 which includes the modulation of the baseband data toIF frequencies and the demodulation of the IF frequencies to basebanddata. The Station Control 14 controls and coordinates the communicationsactivities of the Land Earth station 2.

The Mobile Earth station 4 includes a Mux & Channel Unit 13a, a SIU 10a,a Digital Voice Codec 12a and a Station Control 14a as in the Land EarthStation 2. The Mobile Earth Station does not include a FacilitiesAssignment Processor since, as indicated above, the STU-III terminalsand POTs are directly connected to SIU 10a and Digital Voice Codec 12a,respectively.

Each of the Land Earth Station 2 and Mobile Earth Station 4 includes aSecure Interface Unit (SIU) 10, 10a whose primary function is thedemodulation and remodulation of secure traffic for transmission overthe low-rate mobile satellite circuits. The SIU is located at mobile andland earth stations, where it directly, or indirectly, interfaces withthe end-user secure terminals via a 2-wire connection.

Secure communications, as supported by the network system of the presentinvention, are addressed in the Future Secure Voice System: SignalingPlan-Interoperable Modes, FSVS-210, Revision F, Mar. 31, 1992, NationalSecurity Agency and in Future Secure Voice System: Terminal PerformanceSpecification. FSVS-220, Revision B, Feb. 26, 1988, National SecurityAgency which references are incorporated herein by reference. In thenetwork system of the present invention, only the non-encrypted parts ofthe Secure Terminal unit-version III (STU-III) FSVS signaling plan aredealt with. Encrypted traffic is fully provisioned, but neitherinterfered nor decrypted in the system.

The overall system can be better understood if reference is made to thestate transition diagram (FIG. 2) of a secure (STU-III) call comprisingseven phases which may be defined as follows:

Phase ST-A: Idle (terminal is on-hook)

Phase ST-B: Clear (plain old telephone) mode;

Phase ST-C: Initial call modem training

Phase ST-D: Variable exchange

Phase ST-E: Crypto synchronization and/or resynchronization;

Phase ST-F: Secure traffic; and

Phase ST-G: Call interruption.

During phases ST-D, ST-E, and ST-F, a transparent digital connection(digital pipe) is provided by the SIU 10, 10a to the STU-III signals forthe end-to-end exchange of crypto-synchronization and secure traffic.Only during phase ST-G is the non-encrypted control information(optionally) monitored by the SIU 10, 10a in order to detect callinterruption conditions.

FIG. 3 shows a configuration for implementing a Mobile Earth Station(MES) 4a according to an embodiment of the invention which supportssecure voice or data calls. The configuration shown in FIG. 3 isapplicable to a wide variety of MES terminal implementationsirrespective of the actual manufacturer.

As shown in FIG. 3, the MES 4a includes IF & RF units 20, a Channel Unit22, a Central Call Coding Unit (CCCU) 40, and a Telephone Interface 42.The CCCU 40 includes a programmable Digital Signal Processor 30 whichincludes a Voice Codec Unit 31, a SIU 10 and a Call-Type Discriminator35. The CCCU 40 further includes a Facsimile Interface Unit 36 and aData Interface Unit 38. Facsimile Interface Units are discussed indetail in U.S. application Ser. Nos. 07/754,785 filed Sep. 4, 1991, Ser.No. 07/935,787 filed Aug. 27, 1992 and Ser. No. 07/720,990 filed Jun.26, 1991, which applications are incorporated herein by reference.

Referring to FIG. 3, the IF and RF units 20 and the channel units 22receive the satellite radio frequency signals and then perform thenecessary frequency translation and demodulation of the received signalto the baseband. After this translation and demodulation, the basebandsignal is passed to the CCCU 40 for further processing.

The CCCU 40 may be placed in its entirety within a programmable DigitalSignal Processing (DSP) device, however, the CCCU may be partitionedinto several discrete components. Furthermore, because some of thesecure modes are custom specified, these are likely to be bestimplemented in the same digital signal processor as the voice functions.

For speech signals (plain old telephone, or POT, calls) in theInmarsat-B system, the CCCU 40 activates the digital Voice Codec Unit(VCU) 31 with its input being the analog (or 64 kbit/s PCM digital)voiceband signal and its output being a 16 kbit/s uncoded,parametrically represented baseband signal. For the Inmarsat-M system,the output is a 6.4 kbit/s encoded, parametrically represented basebandsignal.

For facsimile signals, the CCCU 40 activates the facsimile interfaceunit (FIU) 36 whose purpose is the demodulation of facsimile signals andprotocol handling. Its input is the analog voiceband data facsimilesignal and its output is an error protected baseband data facsimilesignal. Any rate adaptation, which may be required due to the manydifferent modems employed during a facsimile transmission, is performedby the FIU 36.

For voiceband data signals, the CCCU 40 takes a form similar to that ofthe facsimile interface: its input is an analog voiceband data signaland its output is a (optionally error protected) baseband data signal.Data are, thus, handled by the Data Interface Unit (DIU) function 38,which may reside within one or more Large Scale Integrated (LSI)components.

For secure calls, the CCCU 40 activates the Secure Interface Unit (SIU)10 whose input is the analog voiceband data signal and its output is anerror protected (coded) 16 kbit/s baseband data signal. When securevoice (e.g., STU-III) terminals operate in the POT mode, only the VoiceCodec Unit 31 of the CCCU 40 is active (the SIU 10 is thus disabled).The SIU 10 may reside within the same programmable component as theVoice Codec 31 because of the programmability of that component, as wellas the possible alternation between the secure and clear modes ofcommunication. Some implementations may, however, partition the SIU 10over several physical components.

The interworking between the different units of the CCCU 40 iscontrolled by the call-type discriminator (CTD) 35. This discriminatormonitors and analyzes the status of the incoming and outgoing signalscontinuously so as to detect changes in the type or spectralcharacteristics of signals being transmitted while a call is inprogress. That is, POT, facsimile, data and secure voice have uniquecharacteristics which allows the CTD to discriminate a call in order toactivate the appropriate one of the Voice Codec Unit 31, SecureInterface Unit 10, Facsimile Interface Unit 36 and Data interface Unit38. Unlike other elements of the CCCU 40 that only operate one at a time(i.e. their operation is mutually exclusive), the CTD 35 always operatesin parallel with one of the units. The CTD 35 may be implemented withinthe programmable VCU/SIU function, or may be implemented as a separateelement.

The digital signal that has been processed by one of the units VCU 31,SIU 10, FIU 36 or DIU 38 of the CCCU 40 will then be converted into ananalog signal at an output interface (not shown) of the CCCU 40.

The converted analog signal will then be placed on the four-wire analogvoiceband bus which is sent to the Telephone Interface 42 which convertsthat signal to a two-wire interface which will then be connected to thetelephone, STU-III terminal, modem or a piece of fax equipment (i.e.,any component that is compatible for connection to a standard telephonecircuit).

As shown in FIG. 3, the output of the Telephone Interface 42 is coupledto a FAX machine, POT, STU-III or Modem via a Multiplexer 45. TheMultiplexer 45 facilitates detection of which service is supposed to beoperating at any one point of time. In other words, since for example, afax call and a voice conversation can not occur simultaneously on thesame channel, the Multiplexer 45 selects only one of those calls at anygiven time. In the embodiment shown in FIG. 3, the Multiplexer 45includes first and second ports for connecting end-user terminals. Thefirst port 1 will be used for normal telephony, facsimile, and securevoice. The second port 2 will be used for voiceband data. Additionalports may be provided for voiceband data, if necessary, in order topermit the reliable detection and accommodation of more types ofvoiceband data modems. This access separation into two ports isconsidered desirable because of the great variety that exists betweenthe call set-up signaling used by different types of voiceband datamodems (thus distinguishing between different modem types andsecure/facsimile is a considerable problem by itself and should behandled separately). By separating the ports into two, the secure voiceterminal problem can thus be confined into a problem of distinguishingbetween voice, facsimile and secure traffic only.

For the voice/facsimile/secure voice port 1, it is possible to assignseparate service addresses to each type of terminal. This may, in fact,occur for facsimile (at least during the initial phases of the serviceprovisioning). In this case, the call discrimination problem reduces toone of distinguishing between POT voice and secure transmissions only.Such a solution does not conflict with the approach taken here; it ismerely a sub-set.

For voiceband data, sophisticated signal discrimination techniques knownin the art may be necessary to distinguish between different voicebanddata types. This is because voiceband data modems use partially commoncall set-up signaling procedures which complicates the reliabledetection of different types of modulation schemes. One approach is toassign separate physical MES entry ports to each type of modem. Anotherapproach is to use separate service addresses. Neither of theseapproaches is very desirable, however, and the use of sophisticatedsignal discrimination techniques may thus be inevitable. In all cases,however, separation of modem traffic from voice, facsimile, and securetransmissions is highly desirable in order to eliminate possible adverseinteractions. Although this differentiation appears to be somewhatrestrictive, it may lead to significantly simpler implementations,particularly if support of additional types of voiceband data terminalsis required in the future. Furthermore, further study may identifyvoiceband modem types which can be guaranteed not to interfere withvoice/facsimile/secure voice call detection, in which case selectedmodem types can be allowed to access port 1.

In the various embodiments it is assumed that voice, facsimile, andsecure voice are addressed by a single access code (terminal number).All traffic, facsimile and secure calls are thus routed to port 1, by anoptional private branch exchange (PBX) or multiplexer 45, as shown inFIG. 3. Voiceband data are routed to a separate port for the reasonsmentioned above.

The Multiplexer 45 selects one of the two ports in accordance with acontrol line from the CCCU 40. For example, the CTD 35 would monitor thecharacteristics of the data that is being received by the CCCU 40, andif those characteristics indicate that data is being supplied, then thisdetermination by the CTD 35 would cause the CCU 30 to activate the DIU38 and to cause Multiplexer 45 via the control line to select port 2 forconnection to the Modem.

The configuration shown in FIG. 3 assumes that each mobile earth stationincludes at least a Central Call Coding Unit (CCCU) whose function is toencode input signals and to convert them into a format suitable forfurther processing by the earth station's channel unit. It is alsoassumed in the FIG. 3 configuration that only one channel unit isprovided (per input circuit). However, those skilled in the art willunderstand that this is not a restrictive assumption under theconfiguration disclosed herein. Specifically, multichannelconfigurations are not expected to impact the protocols describedherein.

The central concept in the SIU architecture is the demodulation ofsignals in the secure terminal-to-satellite direction and there-modulation of signals in the satellite-to-secure terminal direction.This is accomplished by a bank of modulators and demodulators, asillustrated in FIG. 4. Also included in the SIU function are thenecessary GPA and GPC STU-III scramblers and descramblers (used duringthe V.26 modem training phase only), a process controller which providesfor rate adaptation and protocol packetization, an error correctingprocess, and an elastic buffering process. Finally, an echo canceller isalso provided to permit full-duplex communication to be accommodatedover two-wire network access points.

Implementation of the SIU architecture relies heavily on the basicconcepts used in modern modem technology. The digital output of aterminal equipment is modulated to form an analog signal for easiertransmission over communication lines (such as the Public SwitchedTelephone Network or PSTN); conversely, an analog signal is demodulatedto change it back to a digital signal. As an analog modem signal istransmitted over the PSTN, it often encounters imperfections within thenetwork which often makes it difficult for the receiver to correctlydecode the transmitted information. The imperfections which are mostprevalent in the PSTN network are (1) envelope and delay distortion, (2)listener echo, (3) noise impairments such as phase jitters, frequencyoffset, phase hits and gain hits. In most high-performance modems, echocancellers and channel equalizers are used to increase the data ratecapacity of an analog telephone circuit by eliminating the noise addedby the PSTN.

Referring to FIG. 4, the SIU includes an Echo Canceller 50 and a ChannelEqualizer 52 to increase the data rate capacity and eliminate noiseadded by the PSTN. After the network's impairments have been removedfrom the analog signal, the Demodulators 54 convert the analog signal toa digital signal. After the analog signal has been demodulated, it isthen processed by a De-Scrambler 56. Under normal usage, data processingequipment occasionally produces data patterns (like a long string ofones) that can cause the receiving modem problems (usually in theclocking circuitry). As the De-Scrambler 56 in the receiving modemchanges the data back to its original pattern, Elastic Buffers 58 areused to absorb clock differences between the satellite channel and thesecure end-user terminals received (or transmitted) over the telephonecircuit. The receive side Multiplexer 60 is used to select whether theinformation is user data to be transmitted over the satellite channel orsystem control information from the Message (Protocol Control)Packetization process 62 provided by the process controller 63 to betransmitted over the satellite channel. Finally, Error Protection(Coding) 64 is applied to the data which is to be transmitted in orderto protect it from satellite noise impairments.

The inverse operations of the Error Protection Coding 64, Packetization62, Multiplexer 60, Elastic Buffering 58, De-Scrambler 56 andDemodulators 54 are provided by the Error Decoding 65, PacketDisassembly 70, Demultiplexer 66, Elastic Buffering 68, Scrambler 72 andModulators 72, respectively.

It is noted that the traffic supported by the system shown in FIG. 4 isneither decrypted, nor descrambled (with the exception of certain modemtraining segments), nor source decoded. Consequently, the encryption,decryption and STU-III voice coding functions are notably andintentionally absent from the SIU shown in FIG. 4.

Not shown in FIG. 4 is a BCH decoder which may have to be implemented ifcall interruptions are handled using the optional Message Identifier(MID) interpretation procedures addressed below.

According to the embodiments disclosed herein, unless otherwisespecified, the SIU performance characteristics comply with thoseapplicable to STU-III terminals, as specified in FSVS specifications 210and 220 cited above. For example, SIU echo canceller specifications,transmit signal levels and receive signal operating characteristicscomply with FSVS 210 and 220, unless indicated otherwise.

Secure Call Transport Channel

For practical reasons it is desirable to employ the same type of channeltransport mechanism for both voice and secure traffic. This approach wasadopted because it facilitates the use of multiple in-call modificationrequests which may be initiated by the end-user, or dictated by thecharacteristics of the channel, such as when:

A user requests a return to the clear (POT) mode of communication;

A severe fade causes modem training to fail thus requiring a return tothe clear mode followed by modem retraining;

A severe fade causes a burst of uncorrectable errors, eventually leadingto loss in cryptographic synchronization; and

A data channel synchronization discrepancy (slip) causes cryptographicsynchronization loss.

If the voice channel can be used for both voice and securetransmissions, this will also imply that signaling units need not beemployed. This permits the following advantage to be realized:

The channel's full-duplex continuous availability can be guaranteed.This is because the channel unavailability associated with signalingunit channel-type modifications is no longer relevant (which can belonger than 500 ms). This makes it both possible and very convenient toconvert from voice to secure transmissions and vice versa, at any timeduring a secure call.

As set forth below an error correcting code is defined for interoperablemode transmission of secure traffic at an end-user signaling rate of 2.4kbit/s. As also set forth below, error correcting codes (including arate 3/4 error correcting code) is defined for alternate modetransmission of secure traffic at an end-user signaling rate of 4.8kbit/s. Error correcting is applied upon entering phase ST-C (as definedin FIG. 2 and also addressed later).

Error Coding for 2.4 kbit/s Secure Traffic

The general characteristics of the error correcting code to be appliedto the 2.4 kbit/s end-user traffic during the interoperable mode ofcommunication are as follows (see FIG. 5(a) ):

Code rate=3/8;

Channel bit-rate=6400 bit/s;

Code Type=Augmented convolutional;

Constraint length K=7;

Code generator polynomials=133, 171 (octal);

Phase ambiguity resolution=Unique word (e.g., as defined in Inmarsat-MSDM, Section 3.2.2.9);

Modulation method=Offset QPSK;

Modem filter=60% Rolloff;

Demodulator detection=8 level (3 bits) soft decision; and

Code flushing bits=6 "zeros" following last bit of data fed into theencoder (e.g., as defined in Inmarsat-M SDM, Section 3.2.2.10)

It is noted that this error correcting code is also applied to thesecure protocol control packets exchanged between SIUs (defined later).

The following sets forth another embodiment for error coding for 2.4kbit/s secure traffic.

Before 2.4 kbit/s secure traffic can be channel (error) encoded, it israte-adapted to 3.2 kbit/s as follows:

Referring to FIG. 5(b), each voice field, which consists of 320 bits,shall be partitioned into 3 sub-fields as follows. The first sub-field,comprising 48 bits, shall be occupied by the demodulated, 2.4 kbit/ssecure traffic. The first bit in this sub-field that follows the lastbit of the 7 bit dummy field shall correspond to the first bit receivedover the incoming telephone channel. The second sub-field, comprising 16bits, shall be filled by a reserved data pattern. The third sub-field,comprising 256 bits, shall be used by the rate 1/5 error code definedbelow. It is noted that the first two sub-fields, comprising 48+16=64bits, correspond to a net transmission rate of 3.2 kbit/s.

It is noted that only one of the four voice fields (and pertinentsub-band signaling and dummy fields) associated with a signal unit isshown in this figure. It is also noted that the 16 bits in the secondsub-field shall be filled by zeros. This pattern applies to this versionof the protocol only. In future protocol versions, this sub-field may beutilized for enhanced services.

The generator polynomials for the diffused self-orthogonal errorcorrecting code of rate 1/5 to be applied to the 2.4 kbit/s end-usertraffic during the interoperable mode of communication are as follows:

G₁ (D)=1

G₂ (D)=1+D.sup.λ +D¹⁰λ+3 +D¹²λ+4

G₃ (D)=1+D²λ +D¹³λ+5 +D¹⁴λ+7

G₄ (D)=1+D³λ +D⁷λ +D¹⁶λ+8

G₅ (D)=1+D⁵λ +D⁸λ+1 +D⁹λ+2

Code rate=1/5,

Channel bit-rate=16000 bit/s,

Code Type Diffuse Self-Orthogonal,

Code Interleaving Degree (λ)=12,

Processing Delay=16.7 ms.

It is noted that this error correcting code is also applied to thesecure protocol control packets exchanged between SIUs.

Error Coding for 4.8 kbit/s Secure Traffic

The general characteristics of the error correcting code to be appliedto the 4.8 kbit/s end-user traffic during the alternate mode ofcommunication according to one embodiment are as follows (see FIG.6(A)).

Code Rate=3/4

Channel Bit-Rate=6400 bit/s

Code Type=Punctured Convolutional,

Constraint Length K=7; and

Code generator polynomials 133, 171 (octal);

Phase ambiguity resolution=Unique word (e.g. as defined in Inmarsat-MSDM, Section 3.2.2.9);

Modulation method=Offset QPSK;

Modem filter=60% Rolloff;

Demodulator detection=8 level (3 bits) soft decision; and

Code flushing bits=6 "zeros" following last bit of data fed into theencoder (e.g., as defined in Inmarsat-M SDM, Section 3.2.2.10)

Convolutional codes are used to protect digital data from satellitechannel errors, as previously mentioned. FIGS. 5(A) and 6(A) showshift-register circuits which generate a rate 3/8 convolutional code and3/4 convolutional code, respectively. Input bits are clocked into therespective circuits from the left. After each input is applied the coderoutput is generated by sampling and routing the outputs of the twomodulo-2 adders (exclusive-OR). A convolutional code is defined by thenumber of stages in the shift register, the number of outputs (i.e., thenumber of modulo-2 adders), and the connections between the shiftregister and the modulo-2 adders. The state of the encoder is defined tobe the contents of the shift register and is completely determined bythe previous information bit inputs.

The output of the upper modulo-2 adder is the product of the degree-2polynomial G1(D) 1+D² +D³ +D⁵ +D⁶ and the lower modulo-2 adder is theproduct of the degree-2 polynomial G2(D) 1+D+D² +D³ +D⁶ (FIG. 5(A)). Asimilar analysis can be performed on the upper and lower degree-2polynomials of FIG. 6(A). For a given clock period, one bit of inputwill generate two bits of output. For the rate 3/8 augmentedconvolutional coder generator certain designated coder output bits willbe repeated to generate the desired output rate, as indicated in FIG.5(A). For the rate 3/4 punctured convolutional code generator, certaindesignated coder output bits will be deleted to generate the desiredoutput rate, as indicated in FIG. 6(A).

The following sets forth an additional embodiment for error coding for4.8 kbit/s secure traffic.

Before 4.8 kbit/s secure traffic is channel (error) encoded, it israte-adapted to 5.3 kbit/s as follows:

Referring to FIG. 6(b), each voice field, which consists of 320 bits, ispartitioned into 4 sub-fields as follows. The first sub-field,comprising 96 bits, is occupied by the demodulated, 4.8 kbit/s securetraffic. The first bit in this sub-field that follows the last bit ofthe 7 bit dummy field shall correspond to the first bit received overthe incoming telephone channel. The second sub-field, comprising 10bits, shall be filled by a reserved data pattern. The third sub-field,comprising 212 bits, shall be used by the rate 1/3 error code definedbelow. It is noted that the first two sub-fields, comprising 96+10=106bits, correspond to a net transmission rate of 5.3 kbit/s. The finalfield is uncoded and is used only to fill the remaining 2 bits of thevoice field.

It is noted that only one of the four voice fields (and pertinentsub-band signaling and dummy fields) associated with a signal unit isshown in this figure. It is also noted that the 10 bits in the secondsub-field shall be filled with zeros. This pattern applies to thisversion of the protocol only. In future protocol versions, thissub-field may be utilized for enhanced services.

The generator polynomials for the diffused self-orthogonal errorcorrecting code of rate 1/3 to be applied to the 4.8 kbit/s end-usersecure traffic during the interoperable mode of communication are asfollows:

G₁ (D)=1

G₂ (D)=1+D+D.sup.λ+2 +D⁴λ+4 +D⁸λ+5 +D¹⁰λ+10

G₃ (D)=1+D² +D²λ+2 +D⁶λ+4 +D⁷λ+4 +D⁹λ+7

Code rate=1/3,

Channel bit-rate=16000 bit/s

Code Type Diffuse Self-Orthogonal,

Code Interleaving Degree (λ)=15

Processing Delay=11.1 ms.

Call Discrimination

In identifying a secure call, a two layered process needs to beconsidered. First, the calls are grouped into voice and non-voice.Subsequently, non-voice calls are characterized as either facsimile orsecure.

Voice/Non-Voice Call Discrimination

Voice/non-voice call discrimination is accomplished by the call-typediscriminator 35 (FIG. 3) which monitors the type of waveform activitypresented on each direction of signal transmission. Two types of signals(signal₋₋ types) can be used to assist the call-type discriminationprocess:

signal₋₋ type-V: characterized by wideband non-uniform frequencyspectra, non-stationarity, and low transmission levels. This signal typecorresponds to the spectral, energy and temporal characteristics ofspeech signals.

signal₋₋ type-N: characterized by narrowband frequency spectra,stationarity, and high signal transmission levels. This signal typecorresponds to the spectral, energy and temporal characteristics of thefollowing three single-frequency tones: 1100 Hz, 1800 Hz, and 2100 Hz.

A state diagram of the possible call-type transitions is shown in FIG.7. As noted above, the call-type discriminator (CTD) 35 always functionsin parallel with the voice codec, facsimile, or secure interface unit sothat the transition from voice to non-voice call processing can beinitiated at any time during a call in-progress. Once the non-voicestate has been entered, return to the voice state is under the non-voicecoding unit (FIU or SIU) control, or extended presence of signal₋₋type-V.

In particular, for secure transmissions, return to the voice mode isunder secure protocol control, or invoked by the detection of extendedperiods of signal₋₋ type-V activity (which includes idle, or no signalenergy, on the telephone line) as will be discussed later. For facsimiletransmissions, return to the voice mode is under protocol control.

Non-Voice/Secure Call Discrimination

Full-duplex secure calls are always initiated by transmission of an echocanceller or echo suppressor disabling (ECSD/ESD) 2100 Hz tone(thereafter referred to for simplicity as ECSD, unless otherwisenecessary). The full-duplex secure call establishment process over thePublic Switched Telephone Network (PSTN) is illustrated in FIG. 8.

Half-duplex secure calls are always initiated by the transmission of apseudo 1800 Hz tone (often denoted as P1800 Hz, or simply P1800) in theinitiator-to-responder direction. The half-duplex secure callestablishment process is illustrated in FIG. 9.

When group 3 facsimile terminals are configured in the automatic mode ofcommunication, calls are always initiated by transmission of an 1100 Hzcalling tone (CNG) and then followed by a 2100 Hz called stationidentification signal (CED) in the reverse direction of transmission.Facsimile terminals are considered to be in the automatic mode when thedestination address is dialed directly from the key-pad of the end-userfacsimile terminal. When dialing from a facsimile terminal is done usinga separate telephone set, non-automatic call establishment occurs. Inthis case the use of CNG is optional, although all modern facsimileterminals appear to be using the CNG tone even when not dialingautomatically.

The basic protocol for the establishment of facsimile calls over thePSTN is shown in FIG. 10.

From FIGS. 8-10 it can be seen that when an 1100 Hz tone is detected, atransition from the voice codec unit to the facsimile interface unit canbe initiated. Alternatively, when the 2100 Hz tone is detected (whichhas not been immediately preceded by an 1100 Hz tone) or the 1800 Hztone is detected, a transition from the voice codec unit to the secureinterface unit can be initiated. This is shown by the state transitiondiagram shown in the lower part of FIG. 7.

Secure/Facsimile Call Fall-Back

In order to accommodate the possible incorrect call routing that mayarise if older group 3 facsimile terminals (not employing 1100 Hz CNGtransmission) are mistaken for secure terminals, an escape mechanism hasbeen provided as set forth below so that a call can be converted fromsecure to facsimile if signal₋₋ type₋₋ A condition is present: signal₋₋type-A:

Characterized by 2100 Hz activity, followed within 55 to 95 ms by V.21(1750 Hz) carrier activity (in the same direction of transmission)without any activity being present in the return direction oftransmission for the duration of the 2100 Hz and V.21 signals.

The implication of this service-mode conversion is that the firstfacsimile terminal-originated CED/DIS signal pair will be corrupted anda second signal pair will be transmitted from the called terminal afterapproximately 3 seconds. After the second CED/DIS pair has beentransmitted it can be expected that facsimile call establishment willprogress satisfactorily through phases A and B of the CCITTRecommendation T. 30 protocol as described in CCITT Recommendation V.32,"Procedures for Document Facsimile Transmission in the General SwitchedTelephone Network", Blue Book, Melbourne, November 1988.

Call Establishment

The establishment of secure voice calls addressed below, and itsinteraction with voice and non-voice call set-up procedures isidentified.

Call establishment in three different cases is addressed. These casesare: full-duplex interoperable, full-duplex alternate, and half-duplexinteroperable.

Full-Duplex Interoperable

As indicated above, upon detection of a 2100 Hz tone, the transitionfrom the POT voice to the secure mode can, and must, be initiated. Thistransition must satisfy the following requirements:

The 2100 Hz ECSD tone (as seen by the responding secure end-userterminal) must not be disrupted severely by the transition process;

The channel connectivity, on an end-to-end basis, must be maintainedduring the CCCU (central call coding unit) transition process so thatsignal activity following the ECSD tone can be transmitteduninterrupted; and

The transition must be completed within 1 second of the initiation ofthe ECSD tone so that the next signal (phase modulated 1800 Hz) can becorrectly handled by the SIU; and

Finally, the transition must be completed quickly after theestablishment of the ECSD tone (i.e. within 400 ms), so that any phasechanges can be reliably conveyed on an end-to-end basis to networkequipment located in the proximity of the responding end-user secureterminal.

The call set-up approach (refer to FIG. 11 for end-to-end PSTN protocoland to FIG. 12 for end-to-end protocol in the presence of Inmarsat-Bearth-station equipment,) that can satisfy these requirements, and doesnot require the VCU to reliably preserve ECSD phase transitions is thefollowing:

Upon detection of the ECSD tone, the responding earth station'scall-type discriminator recognizes signal₋₋ type₋₋ N activity followedby 2100 Hz activity associated with the initiation of a secure call andinvokes the secure interface unit.

The SIU process is commenced by logically disconnecting the voice codecunit (VCU) from the end-to-end path by instructing the secure interfaceunit (SIU) to commence transmission of the 2100 Hz tone. In this mannertransmission of the 2100 Hz tone to the responding end-user secureterminal continues without a gap in signal energy (although with thepossible introduction of a single phase change due to the transitionprocess).

Coincident with this transition from the VCU to the SIU, the SecureTransmission (ST) Protocol at the responding earth station (forwarddirection) takes over. At this time the 6400 bit/s or 16k bit/s channelcontinues to employ the voice codec in the return responderSIU-to-initiating SIU (return) direction.

Once the forward ST protocol takes over, all further initiating earthstation-to-responding earth station communication is conducted in-band,by means of packets. In particular, the transmission of the non-phasereversed 2100 Hz tone from the responding SIU to the responding secureterminal continues until instructed otherwise from the initiating earthstation end by means of an appropriate packet. Shortly after theresponding earth station call-type discriminator forced the VCU-to-SIUtransition, the initiating earth station's call-type discriminatorrequests the initiating VCU to be logically disconnected and theinitiating SIU to be logically inserted into the channel.

The initiating SIU becomes operational and sends an in-band controlpacket to the responding earth station thus confirming that the STprotocol has now been fully established on an end-to-end basis in theforward direction. On completion of the last bit of this packet, one ofthe error correcting codes defined above is applied to all data sent inthe initiating SIU-to-responding SIU direction.

Within 50 ms of the reception of the (forward) end-to-end ST protocolestablishment packet from the initiating SIU, the responding earthstation will logically disconnect the VCU and connect the SIU in thereturn direction. This is followed by the transmission of an end-to-endST protocol establishment packet in the channel thus confirming that thereturn ST protocol has now been fully established on an end-to-end basisin the return direction. On completion of the last bit of this packet,one of the error correction codes defined above is applied to all datasent in the responding SIU-to-initiating SIU direction.

To ensure that the ST protocol is set-up before the initiator's 2100 Hzphase transitions and/or the responder's 1800 Hz tone are transmitted,the VCU-to-SIU transition process shall be completed in accordance withthe following call set-up timers:

Disconnection of the VCU and connection of the SIU at the respondingearth station in the forward (outgoing) direction: Within 250 ms fromdetection of the ECSD tone (at the responding earth station);

Disconnection of the VCU and connection of the SIU at the initiatingearth station both in the forward and return (incoming and outgoing)directions: Within 275 ms (but no earlier than 250 ms) from detection ofECSD tone (at the initiating earth station);

Transmission of the End-to-End ST Protocol Establishment Control Packetin the forward direction, denoted by CEP(FDX/fwd): Within 25 ms from thedisconnection of the VCU and connection of the SIU (at initiating earthstation).

Disconnection of the VCU and connection of the SIU at the respondingearth station in the return (incoming) direction: Within 50 ms ofreceipt of last bit of CEP(FDX/fwd) at the responding earth station;

Transmission of the End-to-End ST Protocol Establishment Control Packetin the return direction, denoted by CEP(FDX/rtn) : Within 25 ms from thedisconnection of the VCU and connection of the SJU in the respondingearth station's return direction.

A few items should be noted:

First, it should be noted that the secure call can be established byeither land, or mobile earth stations.

Second, since the entire process is initiated by the detection ofin-band signals, the secure voice transmission phase can be enteredwithout interfering with Inmarsat's access control and signalingequipment (ACSE) protocols. In addition the transmission of channel-typeconversion signaling units (as defined in Inmarsat's System DefinitionManuals) is not required. Thus, secure call establishment can be madetransparent to the Inmarsat system, provided that both land and mobileearth station equipment implement the SIU and CTD functional blocks.

Following the VCU-to-SIU change, and while the ECSD tone is beingtransmitted from the initiating terminal, the 2400 bit/s data channelshall be filled with zeros. The responding earth-station willautomatically regenerate a 2100 HZ tone with phase reversals asdesignated in the ISVS specification until a subsequent secure protocolpacket has been received indicating regeneration of another signal.

It is noted that when no signal energy is detected, the 2400 bit/s datachannel (prior to coding and rate adaptation) shall be filled-in withthe 00 di-bit sequence. In particular, this implies that this sequenceshall be transmitted from the responder's SIU to the initiator's SIUupon completion of the CEP(FDX/rtn) until signal activity (P1800 Hz) isdetected in the responder-to-initiator direction.

Full-Duplex Alternate

No additional considerations apply. Call establishment and transition tothe secure protocol communication mode is identical with the full-duplexinteroperable mode.

Half-Duplex Interoperable

As indicated above, upon detection of the pseudo 1800 Hz (P1800) tonethe voice-to-half-duplex secure mode of communication transition must beinitiated. This transition must satisfy the following requirement:

The P1800 Hz tone (as seen by the responding secure end-user terminal)must not be disrupted severely by the transition process; and

The transition must be completed quickly after the establishment of theP1800 Hz tone, so that follow-on P1800 Hz signal activity can bereliably conveyed (regenerated) by the responding secure terminal.

One call set-up approach (refer to FIG. 13 for end-to-end PSTN protocoland to FIG. 14 for end-to-end protocol in the presence of Inmarsat-M orInmarsat-B earth-station equipment) that can satisfy these requirementsis the following:

Upon detection of the P1800 Hz tone, the responding earth station'scall-type discriminator recognizes signal₋₋ type₋₋ N activity followedby (1800 Hz) narrowband activity which is associated with the initiationof a secure call and invokes the secure interface unit.

The secure interface unit (SIU) process is commenced by logicallydisconnecting the voice codec unit from the end-to-end path and byinstructing the SIU to commence transmission of the P1800 Hz tone. Inthis manner transmission of the P1800 Hz tone to the responding end-usersecure terminal continues without a gap in signal energy (although withthe possible introduction of a single phase change due to the transitionprocess).

Coincident with this transition from the VCU to the SIU, the SecureTransmission (ST) Protocol at the responding earth station takes over.At this time the 6400 or 16K bit/s channel continues to employ the voicecodec in the return responder SIU-to-initiating SIU (return) direction.Once the ST protocol takes over, all further initiating earthstation-to-responder earth station communication is conducted in-band,by means of packets. In particular, the transmission of the P1800 Hztone from the responding SIU to the responding secure terminal continuesuntil instructed otherwise from the initiating earth station end bymeans of an appropriate packet. Shortly after the responding call-typediscriminator has forced the VCU-to-SIU transition, the initiating earthstation's call-type discriminator requests the initiating VCU to belogically disconnected and the initiating SIU to be logically insertedinto the channel.

The initiating SIU becomes operational and sends an in-band controlpacket to the responding earth station thus confirming that the STprotocol has now been fully established on an end-to-end basis in theforward direction. On completion of the last bit of this packet, one ofthe error correcting codes defined above shall be applied to all data(after adaptation) sent in the initiating SIU-to-responding SIUdirection.

Within 50 ms of the reception of the (forward) end-to-end (half-duplex)ST protocol establishment packet from the initiating SIU, the respondingearth station will logically disconnect the VCU and connect the SIU inthe return direction. This shall be followed by the transmission of amessage receipt confirmation in the return direction.

The VCU-to-SIU transition process shall be completed within 300 ms ofthe onset of the P1800 Hz tone. In particular the following call set-uptimers apply:

Disconnection of the VCU and connection of the SIU at the respondingearth station in the forward (outgoing) direction: Within 250 ms fromdetection of the first bit of the P1800 Hz tone at the responding earthstation;

Disconnection of the VCU and connection of the SIU at the initiatingearth station in the forward and return (incoming and outgoing)directions: Within 275 ms (but no earlier than 250 ms) from thedetection of the P1800 Hz tone (at the initiating earth station);

Transmission of the End-to-End ST Protocol Establishment Control Packet,denoted by CEP(HDX/fwd): Within 25 ms from the disconnection of the VCUand connection of the SIU (at initiating earth station). Upon completionof the last bit of the CEP(HDX/fwd) packet, the 2400 bit/s data channel(prior to coding and rate adaptation) in the forward direction shall befilled with repetitive transmission of the 02 di-bit sequence.

Disconnection of the VCU and connection of the SIU at the respondingearth station in the return (incoming) direction: Within 50 ms ofreceipt of last bit of CEP(HDX/fwd) at the responding earth station;

Transmission of a Half-Duplex Protocol Establishment ConfirmationControl Packet in the return direction, denoted by CEP(HDX/conf):Immediately upon the disconnection of the VCU and connection of the SIUin the responding earth station's return direction. On completion of thelast bit of this packet, the 16000 bit/s channel shall be filled with anall binary "zero" sequence in the responder-to-initiator direction.

Earth Station-to-Earth Station Control

Once the system is in the Secure Transmission (ST) protocol mode,communication between the land and mobile CCCU is accomplished by meansof secure protocol control (SPC) packets.

Secure control packets are required in order to identify:

Idle state (termination of signal activity);

The start of specific phases of modem training (several controls);

The start of the secure message phase; and

Other types of activity (supervisory or otherwise).

The following characteristics of line control packets are explicitly orimplicitly addressed below:

The type and number of different line control states;

The signal buffering which is needed in order to determine the type ofpacket to be generated;

The structure of different packets;

Packet encoding and insertion in the satellite channel;

The packet detection criteria;

Packet decoding and removal prior to transmission to the secure terminalover the telephone line; and

The modes of line control detection failure and associated callconsequences.

Packet Types

Two types of packets are employed. These are used for call establishment(denoted by CEP) and Secure Protocol Control (denoted by SPC) Eachpacket is associated with a field, whose contents indicate the specificuse of the packet. For example, for the forward full-duplex (FDX) callinitiation, the abbreviated packet designation is: CEP(FDX/fwd).

The following packets are defined, as shown in the table of FIGS. 15 (A)and 15(B). To eliminate future extensions in packet length resultingfrom the need to support additional secure terminal features, a numberof spare packets are also defined. These 31 packets are associated withnumbers 32 to 63.

Packet Structure

As indicated in the table of FIGS. 15(A) and 15(B), there are two typesof packets: call establishment packets (CEP types) and secure protocolcontrol packets (SPC types)--numbers 0 to 63). These are addressedbelow.

FDX/HDX Call Establishment Packet Generation

Packets used to establish a secure call (CEP types in the table of FIGS.15 (A) and 15(B) ) are comprised of a three field or part structurewhich is directly associated with two sub-frames in the 6.4 kbit/sbaseband channel.

Referring to the embodiment of FIG. 16(A), the first field of the fourpossible call establishment special packets comprises 384 repetitions ofa binary "one". The second field comprises 92 repetitions of one of thefollowing 4-bit sequences (depending on the nature of the packet):

CEP(FDX/fwd)0001

CEP(FDX/rtn)0101

CEP(HDX/fwd)0110

CEP(HDX/Conf)0111.

The third field comprises 4 repetitions of a 4-bit SIU architectureversion code. The version applicable to this specification is 0000. Notethat the first bit of each field of the call establishment packets isalways aligned with the first bit of the 384-bit voice sub-field.

According to another embodiment of the invention shown in FIG. 16(b),packets used to establish a secure call (CEP types in FIGS. 15(A)-15(B)are comprised of a three part structure which is directly associatedwith four voice fields in the 16 kbit/s baseband channel (see FIG.16(B).

The first part of the four possible call establishment special packetscomprises 320 repetitions of a binary "one" and is always associatedwith the 320 bit voice field 1. The second part comprises 640repetitions of one of the following 4-bit sequences (depending on thenature of the packet):

    ______________________________________            CEP(FDX/fwd)                    0001            CEP(FDX/rtn)                    0101            CEP(HDX/fwd)                    0110            CEP(HDX/Conf)                    0111.    ______________________________________

This part is always associated with voice fields 2 and 3, which togethercomprise the 640 bits required by the second part of the CEP structure.The third part comprises 80 repetitions of a 4-bit SIU architectureversion code. The version applicable to this specification is 0000

Note that the first bit of each call establishment packet is alwaysaligned with the first bit voice field 1

SPC Packet Generation

Secure protocol control packets (SPC numbers 0 to 63, including spares)are encoded differently from the call establishment packets. Thesepackets comprise a four field structure which includes a leading flagfield, an information field, a frame check sequence field, and an endingoctet (flush) field. Each of these fields is encoded as follows:

Leading Flag: Binary 01111110

Information Field: 8-Bit Binary Representation of the Packet NumberListed in column 1 of FIG. 15(B).

Frame Check Sequence: A 16-bit cyclic redundancy check sequence definedbelow.

Ending Octet (Flush): Binary 00000000

A first embodiment of the packet structure is illustrated in FIG. 17(A)and a second embodiment of the packet structure is illustrated in FIG.17(B). Note that the packets are generated at a rate of 2400 bit/s andthen encoded at a channel rate of 6400 bit/s (FIG. 17(A)) or 16000 bit/s(FIG. 17(B)). Consequently, packets are always treated in the same wayas 2400 bit/s secure traffic, that is, they are inserted in the first 48bits of a voice field (FIG. 17(B) embodiment) In this scheme, thepacket's duration, which impacts the additional delay introduced in theend-to-end communication path due to this protocol, is equal to 16.6 ms(approximately) (FIG. 17(A)) or 20 ms (FIG. 17(B)).

Control Packet Information Field

Although it has not been explicitly shown in the secure transmissionprotocol processing diagrams, all SPC packets are transmitted intriplicate for reliability. After re-transmitting the third packet, thechannel will be immediately followed by either data or zeros dependingupon the SPC packet type. The Repeat Packet Number which appears in theSPC Information Field indicates the sequence of the repeated packet. TheSPC repeat packet "count down" begins at number 3 and is terminated byrepeat packet number 1. The bit structure is illustrated in FIG. 17(C).

Control Packet Frame Check Sequence

The FCS sequence is the same as the sequence used in Group 3 facsimilecommunication (CCITT Recommendation, SS 5.3.7). The FCS shall be a 16bit sequence. It shall be the 1s complement of the sum (modulo 2) of:

the remainder of x⁸ (x¹⁵ +x¹⁴ +x¹³ + . . . +x² +x+1) divided (modulo 2)by the generator polynomial X¹⁶ +X¹² +x⁵ +1, and

the remainder after multiplication by x¹⁶ and then division (modulo 2)by the generator polynomial x¹⁶ +x¹² +X⁵ +1 of the content of the frame.

As a typical implementation, at the transmitter, the initial remainderof the division is preset to all binary "1" and is then modified bydivision by the generator polynomial (as described above) on theinformation field; the is complement of the resulting remainder istransmitted as the 16-bit FCS sequence. At the receiver, the initialremainder is preset to all 1s and the serial incoming protected bits andthe FCS,1 when divided by the generator polynomial, will result in aremainder of 0001110100001111 (X¹⁵ through X⁰, respectively) in theabsence of transmission errors.

The FCS shall be transmitted to the line commencing with the coefficientof the highest term.

Packet Encoding and Insertion in the Satellite Channel

As indicated above, the CEP call establishment packets (special numberss1 to s4) are inserted into the satellite channel coincidentally withthe first bit of the 384 bit voice field or the voice field 1. Thisconstraint does not apply to the SPC packets (numbers 0 to 63) which canbe inserted into coincidentally with the first bit of any voice field.

The 768- or 1280-bit call establishment packets are generated at a 6.4or 16 kbit/s rate and are not error protected. The secure protocolcontrol packets are generated at a rate of 2400 bit/s and then encodedin accordance to the rate 3/8 augmented convolutional code defined aboveor encoded in accordance with the 1/5 code defined above.

Signal Buffering

The use of packets involves the introduction of delay in the end-to-endcommunication system. During the generation of the call establishmentpackets the incoming signal (over the telephone line) shall not bebuffered. This does not apply to the generation of the 40- or 48-bitsecure protocol control packets which are ultimately associated with theonset of end-user data transmission.

Packet Detection Criteria

Two sets of rules apply depending on the type of packet being addressed.

Call Establishment Packet Detection

For call establishment packets, these shall be considered to have beenreceived successfully when the following criteria are met:

At least 368 of the 384 bits or 288 of the 320 bits found in the firstfield are all binary "ones"; and

At least 88 of the 92 replicates of the 4-bit codes are identical or atleast 576 of the 160 replicates of the 4-bit codes associated with parts2 and 3 of these packets are identical; and

At least 2 out of the 4 version codes are identical or at least 72 outof the 80 replicated version codes associated with part 4 of thesepackets are identical.

If these criteria are not met, the POT voice mode of operation shall bemaintained. To avoid inadvertent misclassification of encoded voiceframes as secure call establishment packets, the search for thesepackets shall only be undertaken when the CTD 35 (FIG. 3) has indicatedthat signal₋₋ type₋₋ N activity has been detected (i.e. only after a2100 Hz or 1800 Hz tone has been detected).

SPC Packet Detection

As described above, secure protocol packets are used to conveyinformation regarding the initiation of signal activity. For the secureprotocol control packets, these shall be considered to have beenreceived successfully when the FCS frame associated with the informationfield is correct. Since secure protocol packets are transmitted intriplicate, the receiving unit must successfully receive at least one ofthe three repeated secure protocol packets before interpreting the datawithin the information field. If the FCS frame is found to be in errorin all three of the repeated packets, the receiving unit shall revert tothe clear (POT) mode of voice communication.

Version Numbering

If during call establishment an SIU receives from another SIU a validversion code other than "0000" (that is, the 16-bit or 320-bit versionfield or part received contains at least 2, out of 4, or at least 72,out of 80, version codes other than "0000"), the unit shall ignore suchother codes. It is the responsibility of the unit transmitting the codeother than "0000" to revert to version "0000".

Secure Terminal Protocols

As indicated above, once the end-to-end ST protocol has been establishedcommunication between earth stations is accomplished by means of secureprotocol control (SPC) packets and/or baseband in-band transmittedinformation, as appropriate.

Where appropriate, three variations of protocols are distinguished:full-duplex interoperable, full-duplex alternate, and half-duplexinteroperable. However, the basic protocol is described in terms offull-duplex alternate negotiation, since the other two cases cangenerally be handled as sub-sets of this case.

Pseudo 1800 Hz

Shortly after transmission of the ECSD tone from the initiator, theP1800 Hz tone (modulated by a di-bit pattern) is transmitted from theresponder. When the P1800 Hz tone is used to indicate only basic servicecapabilities (interoperable mode), it carries no phase reversals, inwhich case it can be considered to be modulated by di-bit pattern 02.When the P1800 Hz tone is used to indicate the availability of enhancedservice capabilities by the responder secure terminal (alternate mode),three 180₋₋ phase reversals may be present near the beginning of thistone (modulation using di-bits 01). The P1800 Hz tone is processed bythe SIU as follows:

Upon detection of P1800 Hz activity, the responder SIU will transmit tothe initiator SIU a secure protocol control packet SPC(P1800/s) whichwill indicate to the initiator SIU that the 1800 Hz carrier must beturned on in the earth station-to-terminal direction.

Immediately after transmission of the SPC(P1800/s) packet, repeatedtransmission of the 02 di-bit pattern (which is encoded according to oneof the codes defined above) is initiated.

When a phase change has been detected by the responder SIU, the di-bitdefined above shall b& replaced by the 01 di-bit binary pattern which isencoded with one of the codes defined above prior to transmission overthe 6400 or 16000 bit/s baseband channel.

These processes are shown in FIGS. 18(A) and 18(B).

Any other phase transitions (such as the 3202 di-bit pattern thatcharacterizes the end of the P1800 Hz transmission) shall be similarlycoded using one of the codes as defined above.

Note that an SPC packet indicating the termination of the P1800 Hz isnot necessary in this case, as the P1800 Hz signal will be followed by adifferent type of signal with no interruption in signal energy.

It should be further noted that, during alternate mode transmissionssignals transmitted in a direction opposite to that of the P1800 Hz maybe present. For simplicity and clarity of presentation, these are notshown in FIGS. 18(A)-18(B).

Message A

Shortly after termination of the initiator's ECSD tone (within 85+/-10ms), the initiator may transmit an enhanced capabilities 300 bit/ssignal, denoted by MSG A, which is modulated in accordance to Bell Modem103 (modified). This signal will contain a profile of initiator terminalcapabilities including:

Capability for extended echo ranging process;

Capability to operate at 4800 bit/s; and

Capability to operate at 9600 bit/s.

The initiator SIU shall intercept and set to binary zero (if not alreadyset) the bits indicated in FIGS. 19(a), 19(b) and 19(c) so as to disablecertain non-supportable capabilities, such as extended echo ranging.

The ST protocol shall process the MSG A signal as follows:

Upon detection of MSG A signal activity, the initiating SIU shalltransmit to the responding SIU a secure protocol control packet SPC(Bell103/s) which will indicate to the responding SIU that the Bell 103carrier must be turned on in the earth station-to-terminal direction.

Immediately after transmission of the SPC(Bell 103/s) packet, theinitiating SIU shall transmit the associated baseband information at arate of 2400 bit/s (prior to coding). The rate adaptation from 300 bit/sto 2400 bit/s is as follows. Every 300 bit/s binary bit transmitted fromthe secure terminal to the SIU shall be converted into the following setof 8-bits which are then encoded using one of the codes defined abovefor transmission over the 6.4 or 16 kbit/s satellite channel:

Binary Bit 0 converted to Set of 8-Bits: "11101110"!

Binary Bit 1 converted to Set of 8-Bits: "01010101"!

The baseband information shall be unmodified, with the possibleexception of those bits indicated in the tables of FIGS. 19(a), 19(b)and 19(c).

When termination of MSG A occurs, the initiating SIU shall transmit tothe responding SIU a secure protocol control packet SPC(Bell 103/e)which will indicate to the responding SIU that the Bell 103 carrier mustbe turned off in the earth station-to-terminal direction. The potentialelimination of this type of packet is for further study.

These processes are shown in FIG. 20. Note that signals transmitted in adirection opposite to that of MSG A may be present, but for clarity andsimplicity these are not shown in FIG. 20.

Message B

Immediately after termination of the responder's P1800 Hz signal, theresponder may transmit an enhanced negotiated capabilities 300 bit/ssignal, MSG B which is modulated in accordance to Bell Modem 103(modified). This signal will contain the initiator's capabilities whichwere chosen by the responding secure terminal for the remainder of thecall.

Unlike MSG A neither the responding nor the initiating SIU may modifythe contents of this signal. The ST protocol shall process the MSG Bsignal as follows:

Upon detection of MSG B signal activity, the responding SIU shalltransmit to the initiating SIU a secure protocol control packet SPC(Bell103/s) which will indicate to the initiating SIU that its carrier in theearth station-to-terminal direction must be switched from 1800 Hz to theBell 103 carrier.

Immediately after transmission of the SPC(Bell 103/s) packet, theresponding SIU shall transmit the associated baseband information at arate of 2400 bit/s (prior to coding). The rate adaptation from 300 bit/sto the 2400 bit/s is identical to that used for MSG A coding.

When termination of MSG B occurs, the responding SIU may transmit to theinitiating SIU a secure protocol control packet SPC(Bell 103/e) whichwill indicate to the initiating SIU that the Bell 103 carrier must beturned off in the earth station-to-terminal direction. The potentialelimination of this type of packet is for further study.

These processes are shown in FIG. 21. Note that if the alternate mode isnot selected, it may not be necessary to explicitly indicate that theMSG B signal has terminated, since this will be immediately followedwithout any interruption in signal energy by the P1800 Hz tone the startof which will be signaled by means of the SPC(1800/s) packet.

However, because the alternate mode may be selected, an indicationpertaining to the end of the Bell 103 transmission is required. (This issignaled by the transmission of the SPC (Bell 103/e) packet). it is alsonoted that the responder may implicitly derive the termination of theBell 103 transmission through processing of the MSG B contents.

Following the termination of the MSG B signal, one of the followingoptions may be invoked:

The call is set up in an alternate mode; or

The call is set up in the interoperable mode.

Alternate Mode

When a call is set-up in the alternate mode as a result of MSG A and MSGB negotiation, the modem training sequence defined in CCITTRecommendation V.32, "A Family of 2-Wire, Duplex Modems Operating atData Signaling Rates of Up to 9600 bits/s for Use in the GeneralSwitched Telephone Network and on Leased Telephone-Type Lines", BlueBook, Facsimile VIII. 1. pp. 234-251, Melbourne, November 1988, SS 5.4.1and SS 5.4.2 will be employed by the end-user secure terminals. 11 parts(segments) of the training sequence shall be regenerated by the SIUfunction in the satellite-to-secure terminal direction. Theirregeneration, however, is controlled by the exchange of control packetsso that the correct end-to-end timing relationship can be maintained (atthe end-user secure terminals) in a transparent manner. For the purposeof simplifying references to the V.32 sequence in this document, thesequence is partitioned into five segments: three for the responder,designated as R1, R2 and R3 (not to be confused with the V.32 ratesequences which are denoted by R1 and R2) ; and two for the initiator,designated as I1 and I2. This notation is indicated in FIG. 22.

This notation is subsequently used to associate control packets with thetype of training segment to which they relate.

With this definition in mind, a number of packets are exchanged betweenthe two SIUs to specifically trigger initiation of different parts ofeach of the five training sequence segments. The exchange of packets andassociated protocol is indicated in FIGS. 23(a)-23(d). Several itemsneed to be noted from this set of figures:

First, the end-to-end delay, as well as the delay between each earthstation and its near-side secure interface unit can be derived from thistraining procedure and particularly from the exchanges related totraining segments I1 and R1.

Second, when the alternate mode is invoked a set of capabilities must beprecluded from being invoked. To accomplish this, the initiating SIUmust generate a predetermined rate sequence irrespective of the sequencegenerated from the responding secure terminal. This rate sequence isdefined in the Table of FIG. 24.

Although not explicitly noted, the ending sequence E, shall also beencoded using the same profile of capabilities as those indicated in theTable of FIG. 24.

The optional extended echo control training sequence (which may beappended in front of training segment R2) is not supported.

The completion of the modem training sequence is defined when the lastbit of the SPC(V.32/I2, Bl/e) and SPC(V.32/R3, Bl/e) packets has beentransmitted for the initiating-to-responding (forward) andresponding-to-initiating (return) directions, respectively.

Immediately upon completion of the modem training sequences" the Startof Message (SOM) which characterizes the onset of phase ST-D (defined inFIG. 2) shall be entered into. Coincidentally with the first bit of theSOM error code defined above shall be applied. This type of coding shallbe applied for the remainder of the transmission while the call is inphases ST-D, ST-E, ST-F, and ST-G.

Interoperable Mode

As indicated above, it is possible that the interoperable mode isselected as a result of the MSG A and MSG B negotiation (FIGS. 2-5 ofthe PSVS-210 reference). In this case, the remainder of the call set-up(modem training) will proceed differently from that addressed in thealternate mode discussed above.

Responder's Scrambled ones (SCR 1)

Upon termination of the 1800 Hz (phase reversed) tone, the initiator'svoiceband data modem is trained. This is accomplished by the initiator'stransmission of a 4096-bit scrambling sequence (SCR1). The SCR 1sequence always follows the 1800 Hz tone with 0 ms delay (i.e. no gap insignal energy). The last di-bit sequence of the P1800 Hz tone is not0202, but rather 3202.

In the Inmarsat-M and Inmarsat-B secure transmission protocol the SCR 1sequence is not sent across the satellite channel. Instead this will beregenerated by the initiating SIU on the basis of the SCR 1 packet,SPC(SCR1). The protocol to be used in this phase is as follows:

Upon detection of the 1800 Hz modulated 3202 di-bit pattern, theresponding SIU is prepared to accept the SCR 1 sequence and starttraining its own modem. The di-bit 3202 sequence is transmitted acrossthe channel followed by a SPC(SCR1) packet. The possible elimination ofthe SCR1 packet is for further study.

Upon completion of the SCR 1 sequence from the responding secureterminal, the responding SIU will send a secure protocol controlSPC(Idle) packet to the initiating SIUII indicating the termination ofvoiceband data carrier on the telephone line. Implicit determination ofsignal termination is also possible on the basis of the number of SCR1bits received (or regenerated).

The possible elimination of the SPC(Idle) packet is for further study.

These processes are shown in FIG. 25.

Initiator's Scrambled Ones (SCR 1)

Upon completion of the responder's SCR 1 sequence, the initiatingterminal will send a SCR 1 sequence to the responder's voiceband datamodem so that its receiver can be appropriately trained. The initiator'sSCR 1 sequence will always be preceded (with no interruption in signalenergy) by a 2100 Hz ESD or ECSD tone (which may thus include 180₋₋phase reversals). The treatment of the initiator's SCR 1 sequence issimilar to that of the responder's SCR 1 sequence and is defined abovewith respect to the Responder's Scrambled Ones (SCR1).

The initiation of the 2100 Hz ECSD is conveyed by means of packetSPC(2100). Unlike the P1800 sequence, no di-bits are associated with theSPC(2100) packet to indicate phase shifts in the regenerated signal. Theexact ST protocol is as follows:

After completion of the responder's SCR 1 the initiating SIU shallmonitor the incoming telephone line in order to detect the onset of the2100 Hz ECSD tone. When detected, the initiating SIU shall transmit tothe responder's SIU a secure protocol control SPC(2100) packet whichshall signal the responding SIU to turn the 2100 Hz tone on in the earthstation-to-secure terminal direction.

Immediately upon completion of the SPC(2100) packet, the channel isfilled with zeros. The earth-station that receives the PSC92100) packetwill automatically regenerate a 2100 Hz tone with phase reversals asdesignated in the FSVS specification until a subsequent secure protocolpacket has been received indicating regeneration of another signal. Aswith SCR1 sequence, the 2100 Hz tone with phase reversal is notexplicitly sent across the satellite channel.

Instead, the signal is regenerated by the SIU receiving a SPC(2100)packet. In addition, it is assumed in the Inmarsat-M secure transmissionprotocol that a 2100 Hz tone with phase reversals has been detected.

Upon detection of the initiator's SCR 1 sequence, the initiating SIUshall prepare to accept the SCR 1 sequence and start training its ownmodem. The SPC(SCR1) packet shall then be transmitted across the channelto the responding SIU.

Upon completion of the SCR 1 sequence from the initiating secureterminal, the initiating SIU shall transmit a secure protocol controlSPC(Data) packet to the initiating siu, indicating the end of the SCR 1sequence and the start of secure traffic transmission. Thisdetermination is also possible on the basis of the number of SCRI bitsreceived (or regenerated). The possible elimination of the SPC(Data)packet is for further study. Upon completion of the SPC(Data) packet, a2.4 kbit/s digital data pipe (prior to coding) shall be established in atransparent manner for the initiator's terminal.

These processes are shown in FIG. 26.

Responder's Second Set of Scrambled Ones

Upon completion of the initiator's SCR 1 sequence, the responderterminal will send a shorter (704 bit) SCR 1 sequence to the initiator'svoiceband data modem followed immediately by secure data. Unlike theresponder's first SCR 1 sequence, this sequence will not be preceded bya 2100 Hz ESD or ECSD tone. The treatment of the second SCR I sequenceis similar to that of that of the responder's first SCR 1 sequence. Theexact ST protocol is as follows:

After completion of the initiator's SCR 1 the responding SIU shallmonitor the incoming telephone line in order to detect the onset of theresponder's shorter SCR 1 sequence. When detected, the responding SIUshall transmit to the initiating SIU a secure protocol control SPC(SCR1)packet which shall signal the initiating SIU to commence transmission ofthe shorter SCR 1 sequence (appropriately modulated) in the earthstation-to-secure terminal direction.

Upon completion of the SCR I sequence from the responding secureterminal, the responding SIU shall transmit a secure protocol controlSPC(Data) packet to the initiating siu, indicating the end of the SCR 1sequence and the start of secure data transmission. This determinationis also possible on the basis of the number of SCRI bits received (orregenerated). The possible elimination of the SPC(Data) packet is forfurther study. Upon completion of the SPC(Data) packet, a 2.4 kbit/sdigital data pipe (prior to coding) shall be established in atransparent manner for the responder's terminal.

Other Modes

It is possible that a call can be established in other communicationmodes. For example, an interoperable mode (FIGS. 2-3 of the referenceFSVS-210) can be established without use of the MSG A and MSG B signals.This case is merely a sub-set of the procedure defined previously as thecontrol packets needed to fully describe the modem training process havebeen defined above.

This observation also applies to the case of half-duplex transmission,as illustrated in FIG. 27.

Call Failures & Clearing

Once a call has entered the secure transmission protocol phase ST-D thesecure terminal data are no longer interpreted with the exception ofcertain non-data bearing messages. These messages, which are notencrypted, are the following:

Abort;

Release;

Failed Call;

Restart Failed Call;

Idle;

Retrain Request;

Retrain NACK; and

Retrain ACK.

These messages will be detected by the SIU function by confirming thatthey are preceded by the 256-bit "Escapell and 64-bit "Start of Message"non-encrypted segments.

Abort

When this signal is received (which can be assumed to be transmittedfrom the leader's to the follower's secure terminal) the earth stationshall initiate the transition from secure to POT voice transmission(SIU-to-VCU). This transition shall be initiated following the last bitof the Message Identifier (MID) in the following manner:

For the leader's SIU in the leader-to-follower direction: Within 25 Msfollowing the transmission of the last bit of the associated MID field;

For the follower's SIU in the leader-to-follower direction: Within 135ms following the reception of the last bit of the associated MID field;

For the follower's SIU in the follower-to-leader direction: Within 165ms following the reception of the last bit of the associated MID field;and

For the leader's SIU in the follower-to-leader direction: Within 800 ms(but no earlier than 700 Ms) following the transmission of the last bitof the associated MID field.

The application of these rules is illustrated in FIG. 28.

Failed Call

When this signal is received (which can be assumed to be transmittedfrom the leader's to the follower's secure terminal) the earth stationshall initiate the transition from secure to POT voice transmission(SIU-to-VCU). This transition shall be initiated following the last bitof the Message Identifier (MID) associated with a failed call and shallbe accomplished in the exact same manner which is described above forthe Abort condition.

Release

When this signal is received (which can be assumed to be transmittedfrom the leader's to the follower's secure terminal) the earth stationshall initiate the call clear-down procedures. This will be accomplishedby first initiating the POT voice codec unit (SIU-to-VCU transition orrelease/1) and then by clearing the channel as a normal voice call(release/2).

The transition from the SIU to the VCU shall be initiated following thelast bit of the Message Identifier (MID) associated with a releaserequest and shall be accomplished in the exact same manner which isdescribed above for the Abort condition. The channel release proceduresshall be fully compliant with Inmarsat's voice call clearing procedures.

EOM

When this signal is received (which is associated with half-duplex callsand can be assumed to be transmitted from the initiator's secureterminal to the responder's secure terminal) the earth station shallinitiate the transition from secure to POT voice transmission(SIU-to-VCU). This transition shall be initiated following the last bitof the Message Identifier (MID) associated with an end of message statusand shall be accomplished in the exact same manner which is describedabove for the Abort condition. In this case it is noted that the voicechannel is being established in both directions of transmission(full-duplex).

The support of the above non-data bearing messages is illustrated in thestate diagram shown in FIG. 29.

Idle

This message is related to half-duplex transmission when communicatingwith the Key Management Center (KMC).

Retraining Messages

A series of messages such as Retrain Request, Retrain NACK, and RetrainACK are related to retraining of modems.

Carrier Loss

If modem carrier is lost in either direction of transmission duringphases ST-D, ST-E, ST-F, and ST-G (FIG. 2), the secure to POT voicetransmission procedures applicable to EOM shall be followed. Thisrequirement does not apply if carrier is lost (or absent) for less than1 second, or if the carrier is absent in the return direction of a callestablished in the half-duplex mode of communication.

The procedures defined in the previous paragraphs may optionally beby-passed by monitoring the presence or absence of modem carrier inorder to revert to the POT mode of operation. in particular, transitionfrom the secure (Phase ST-F) to the POT mode of operation may be made,if loss of carrier is detected in either direction (full-duplex case) orin the forward direction (half-duplex case) for more than 1 second.

When the call is in the modem training (ST-C) phase, the aboverequirements are modified so that the pertinent timer is increased from1 to 4 seconds.

Echo Control

Secure terminals incorporate the ability to determine path length(round-trip transmission delay) through the use of a ranging process.Since this determination is made at the beginning of the call, care mustbe taken to ensure that this determination is not made over the voicepath. This is because upon call routing to secure demodulatingfacilities the path-delay characteristics will change significantly, andthe delay measured over the voice path will no longer be applicable.

With regard to echo control, the following considerations apply (FIG. 30establishes the echo control reference framework).

First, for the end-user secure terminals there is no far-end echo pathestablished since the end-to-end transmission of secure data will beaccomplished by regeneration of the voiceband data signal by the far-endearth station. It is noted that upon establishment of the demodulatingfacilities the echo path will exhibit an echo loss discontinuity,assuming network stability.

As a result, there is no need for far-end cancellation and associatedecho-ranging for either near-end (E/C 1) or far-end (E/C 4) secureterminal cancellers. For this reason, the echo-ranging option isdisabled by the intervening earth-station network discussed above. Thisdelegates the secure terminal cancellers to canceling near-end echoesonly.

With regard to the two earth-station echo cancellers, differentconsiderations apply in each case.

First, the near-end mobile earth station canceller (E/C 2) will belooking into the near-end secure terminal over a very short delay path(<1 ms). Second, it can be assumed that since the earth station will bepermanently connected to a 4-to-2 wire terminating equipment (or privatebranch exchange), the termination's balancing characteristics can beadjusted sufficiently well so as to minimize the need for echo control.It is anticipated that a return loss of 25 dB can both be easilyachieved may also be acceptable for satisfactory voiceband dataperformance.

If this performance cannot be guaranteed, then an 8-tap echo cancellerable to deliver an echo return loss of 25 dB) should be implementedwithin the MES earth station.

With regard to the far-end land earth station canceller (E/C 3), thiswill be looking into the far-end secure terminal over a longer delaypath (-5 to 400 ms). Unlike the mobile-side equipment, it can no longerbe assumed that the earth station will be permanently connected to a4-to-2 wire terminating equipment and, thus, the termination's balancingcharacteristics will vary from call-to-call. Echo control will thuscertainly be required at the land earth station.

This echo canceller must provision for near end echoes (echo paths withround-trip delays of up to 12 Ms) and far end echoes echo paths withround-trip delays of up to 400 ms).

The echo return loss performance requirement and convergence time forthis canceller must meet the requirements applicable to the STU-IIIterminals themselves (as discussed in the FSVS-220 reference citedabove).

Delay Path Estimation

As indicated above, the extended echo path ranging process is disabledby the SIU. Since, however, the far-end LES-based SIU needs toaccommodate long echo paths, a method is required to permit the far-endecho canceller (E/C 3) to estimate the length or delay of the echo pathbetween the LES and the PSTN-based secure terminal. This can beaccomplished in a number of ways. For interoperable as well as alternatemodes of operation, the echo associated with the onset of the 2100 HzECSD tone at the beginning of the call can be used for this purpose (formobile originated calls only). In addition, for interoperable calls(only), the delay between the end of the responder's SCR1 sequence andthe onset of the initiator's ECSD tone can be used to provide anotherpath-delay estimate. (The delay between the end of the initiator's SCR1sequence and the onset of the initiator's SCR1 sequence can provide yetanother estimate).

For alternate calls (only), the echo path delay can be estimateddirectly from the modem training sequence during segments I1 and R1.These segments are specifically designed to permit this delay to bederived; and the use of packets to trigger the onset of specificsegments of modem training is one of the reasons for exercising thepacketized approach on an end-to-end, rather than local-to-local, modemtraining basis.

Elastic Buffering

In order to compensate for the interruption in synchronicity of theend-to-end data path arising from the demodulation-remodulation process,a signal buffering process is defined. Through this process the secureinterface units (SIUS) absorb clock differences between the satellitechannel and the secure end-user terminals received (or transmitted) overthe telephone circuit by means of slip control (i.e., loss orduplication of data) using an elastic buffer 58,68 (FIG. 4).

The elastic buffer used for slip control shall have a capacity of 576bits (which is equivalent to 120 ms at a 4800 bit/s transmission rateassuming an end-user clock accuracy of 10-4.

When the demodulating SIU commences transmission of the demodulated datastream to the remodulating SIU, the read pointer of the elastic buffershall be reset to the following position in relation to the writepointer:

For 2.4 kbit/s transmissions 144 bits behind

For 4.8 kbit/s transmissions 288 bits behind

This implies that the demodulating SIU will add a nominal 60 ms delay atbeginning of such transmissions, in addition to any other delay (such asprocessing). Care must be exercised to ensure that duringinteroperable/alternate mode establishment an appropriate amount ofdelay is introduced to the secure protocol control packets associatedwith the initiation of the end-users'data.

If the read pointer reaches the write pointer during the transmission ofa signal (as a result of the end-user terminal clock rate being lowerthan the satellite channel clock rate), a slip operation shall beperformed by shifting the read pointer back (i.e. duplication of data),so as to set the read pointer behind the write pointer by the sameamount as that specified above (as appropriate to each of the twoend-user signaling rates).

If the read pointer becomes twice as many bits behind as that specifiedabove (as appropriate to each of the two end-user signaling rates a slipoperation shall be performed by shifting the read pointer ahead by thesame amount as that specified above (i.e. loss of data), so as to setthe read pointer behind the write pointer by the same amount as thatspecified above. This case arises when the end-user terminal clock rateis higher than the satellite clock rate.

Whenever the transmission of a signal on the satellite channel (by thedemodulating SIU) is referred to in the above paragraphs, it shall beinterpreted as the input of the data to the elastic buffer of thedemodulating SIU prior to error coding, and not the actual transmissionto the satellite channel.

The above requirements do not apply to the re-modulating SIUS. At there-modulating SIU, synchronicity shall be maintained by driving thetransmit direction of the modem (outgoing telephone circuit direction)with the clock associated with the receipt of data from the satellitechannel.

Maximum Allowable Processing Delays

In the process of providing elastic buffering, signal packetization, andother types of processing, the SIUs will introduce additional delays inthe end to end communication path. The maximum allowable delays are asfollows:

    ______________________________________    Demodulating SIU Side:    Packet Assembly    20.0 ms    Elastic Buffering  60.0 ms    Processing Delay   20.0 ms    Total Delay       100.0 ms    Remodulating SIU Side:    Packet Disassembly                       40.0 ms    Processing Delay   20.0 ms    Elastic Buffering  60.0 ms    Total Delay       120.0 ms    ______________________________________

It is noted that the current elastic delay specification permitsapproximately 10 minutes of slip-free operation to be supported.

Special Voice Frames

In order to permit the voice encoded channel to support narrowbandsignals, such as the 2100 Hz and 1800 Hz tones used in secure (andother) communications, an enhancement to the utilization of voice framesis defined below.

The Inmarsat-M voice coding algorithm transmits 128 bits of informationwith every 20 ms frame. Each 128 bit frame is divided among 8 codevectors which in the Inmarsat-M SDM are denoted as co to C7- Each ofthese code vectors is generated by error encoding of a correspondingdata vector which is denoted by U0 to U7. In the Tables of FIGS. 31(a),31(b) and 31(c), the format of these vectors are defined.

The remaining seven bits in the 20 ms frame comprising data vector U7are set to binary Ilzeroll.

It is noted that due to physical limits imposed on the value which thespeech fundamental frequency can attain, the first six bits of vector UOwill never equal the decimal value of 48 under error-free conditions.This observation is exploited to indicate to an enhanced voice decoderthat a special signal has been detected. It also noted that if thisframe is received by a receiver not capable of secure operation, thecorrect action will be a frame repeat of the most recently receivedvalid voice encoded frame.

The gain index GD used in the tables of FIGS. 31 is a number in therange of 0 to 256 decimal which corresponds to the level of the receivedtone as follows:

Tone Level=(O-0.17 GD ) dBmO

It is noted that a value of GD=0 corresponds to a signal level of 0 dBmOand a value of GD=256 corresponds to a signal level of -43.52 dBmO.

The four-bit "Tone Index" T, which is used above are defined as theTable of FIG. 32.

Tone Detection Requirements

The following requirements apply to the detection of 1800 Hz and 2100 Hztones.

Maximum voice/tone index transition: 25 ms Upon detection of the onsetof a valid 1800 Hz or 2100 Hz tone, no more that 25 ms of tone shall beencoded and transmitted in the voice mode.

Maximum false alarm rate: 2×10-5

For any non-1800 Hz or 2100 Hz tonal input, the fractional number of 20ms frames transmitted in the tonal index mode shall not exceed thenumber given above.

Other requirements, such as input dynamic range, input signal to noiseratio, frequency tolerances, and amplitude accuracies are to becompliant with the relevant FSVS specifications.

Tone Generation Requirements

The relevant FSVS specifications define requirements that apply to thegeneration of 1800 Hz and 2100 Hz tones. Such requirements include, forexample, the tones'output dynamic range, signal-to-noise ratio,frequency accuracy, and amplitude accuracy.

Having described the invention in detail, those skilled in the art willappreciate that numerous modifications may be made of the inventionwithout departing from its spirit. Therefore, it is not intended thatthe scope of this invention be limited to the specific embodimentsillustrated and described. Rather, it is intended that the scope of theinvention be determined by the appended claims and their equivalents.

What is claimed is:
 1. A communication system, comprising:a firstcommunication terminal for providing interchangeably both secure andnon-secure information carried by analog voiceband signals; a firstprocessing circuit connected to receive analog voiceband signal(s) fromthe first communication terminal and for converting the received analogvoiceband signal(s) into digital baseband data, said first processingcircuit operating interchangeably and automatically on secure andnon-secure informatation carried by analog voiceband signals as a userof the first communication terminal changes between secure andnon-secure information, and said first processing circuit providing thedigital baseband data for transmission; a second processing circuitconnected to receive the transmitted digital baseband data from thefirst processing circuit, and for converting the received digitalbaseband data into analog voiceband signal(s), said second processingcircuit operating interchangeably and automatically on the receiveddigital baseband data to convert said data to analog voiceband signalscarrying the secure or non-secure information provided to the firstprocessing circuit; a second communication terminal for receiving theanalog voiceband signal(s) from said second processing circuit; and asatellite communication link, wherein said digital baseband data istransmitted from said first processing circuit via said satellitecommunication link.
 2. The communication system as defined in claim 1,wherein said digital baseband data has a nominal rate of 9.6 kbit/s orless.
 3. The communication system as defined in claim 1, wherein saiddigital baseband data has a nominal rate of 4.8 kbit/s.
 4. Thecommunication system as defined in claim 1, wherein said digitalbaseband data has a nominal rate of 2.4 kbit/s.
 5. A communication unitcomprising:a communication terminal for providing and receivinginterchangeably both secure and non-secure information carried by analogvoiceband signal(s); a processing circuit connected to receive theanalog voiceband signal(s) from said communication terminal and forconverting the received analog voiceband signal(s) into digital basebanddata, said processing circuit operating interchangeably andautomatically on secure and non-secure information carried by analogvoiceband signals as a user of the communication terminal chancesbetween secure and non-secure information, said processing circuitproviding the digital baseband data for transmission, said processingcircuit also receiving transmitted digital baseband data and convertingthe received digital baseband data into analog voiceband signal(s), saidprocessing circuit operating interchangeably and automatically on thereceived transmitted digital baseband data to convert said data toanalog voiceband signals carrying secure and non-secure information;wherein said processing circuit includes a signal detector fordistinguishing between secure and non-secure information received fromsaid communication terminal.
 6. The communication unit as defined inclaim 5, wherein said communication terminal includes a STU-IIIterminal.
 7. The communication unit as defined in claim 5, wherein saiddigital baseband data has a nominal rate of 9.6 kbit/s or less.
 8. Thecommunication unit as defined in claim 7, wherein said digital basebanddata has a nominal rate of 4.8 kbit/s.
 9. The communication unit asdefined in claim 7, wherein said digital baseband data has a nominalrate of 2.4 kbit/s.
 10. A method for providing communications betweenSTU-III terminals over narrow bandwidth digital circuits,comprising:providing secure and non-secure information carried by analogvoiceband signals from a first STU-III terminal; receiving the providedinformation carried by analog voiceband signal(s) from the first STU-IIIterminal and distinguishing between secure and non-secure informationcarried by the received analog voiceband signals; converting the analogvoiceband signals into digital baseband data; transmitting the digitalbaseband data; receiving the transmitted digital baseband data;converting the received digital baseband data into analog voicebandsignals carrying the secure and non-secure information provided by thefirst STU-III terminal; and providing the analog voiceband signals to asecond STU-III terminal.
 11. The method of claim 10, wherein saiddigital baseband data is transmitted at a nominal rate of 9.6 kbit/s orless.
 12. The method of claim 11, wherein said transmitted digitalbaseband data has a nominal rate of 4.8 kbit/s.
 13. The method of claim11, wherein said transmitted digital baseband data has a nominal rate of2.4 kbit/s.
 14. The method of claim 10, wherein said transmitting stepis performed via a communication satellite link.
 15. The method of claim14, wherein said digital baseband data is transmitted over saidsatellite link at a nominal rate of 9.6 kbit/s or less.
 16. The methodof claim 14, wherein said digital baseband data has a nominal rate of4.8 kbit/s.
 17. The method of claim 14, wherein said digital basebanddata has a nominal rate of 2.4 kbit/s.
 18. A method of establishingsecure communication between a first STU-III terminal system and asecond STU-III terminal system, each of the first and second STU-IIIterminal systems includes a digital voice codec fortransmitting/receiving telephone signals, and a secure interface unitfor transmitting/receiving secure telephone signals, the methodcomprising:transmitting from the first STU-III terminal system apredetermined tone signal; receiving by the second STU-III terminalsystem the predetermined tone signal; and disconnecting the digitalvoice codec and connecting the secure interface unit of the secondSTU-III terminal system in response to receiving the transmittedpredetermined tone.
 19. The method as defined in claim 18, wherein thepredetermined tone signal is one of a 2100 Hz and a pseudo 1800 Hz tonesignal.
 20. The method as defined in claim 18, further comprisingdisconnecting the digital voice codec and connecting the secureinterface unit of the first STU-III terminal system within apredetermined time from the transmission of the predetermined tonesignal.
 21. The method as defined in claim 18, further comprisingtransmitting from the first STU-III terminal system a first controlpacket confirming that a secure protocol has been established by thefirst STU-III terminal system.
 22. The method as defined in claim 21,further comprising transmitting from the second STU-III terminal systema second control packet confirming that a secure protocol has beenestablished by the second STU-III terminal system.
 23. The method asdefined in claim 22, wherein communication between the first and secondSTU-III terminal systems is by a satellite communication link.
 24. Themethod as defined in claim 23, wherein the second control packet istransmitted within 50 ms after the first control packet is received bythe second STU-III terminal system.
 25. The method as defined in claim23, further comprising transmitting from the first STU-III terminalsystem a secure protocol control packet for initiating secure datatransmission.
 26. The method as defined in claim 18, further comprisingtransmitting by the second STU-III terminal system a secure protocolcontrol packet indicating the capabilities of the second STU-IIIterminal system.
 27. A method of establishing a secure protocol in aSTU-III terminal system, the STU-III terminal system including a digitalvoice codec for transmitting/receiving telephone signals, and a secureinterface unit for transmitting/receiving secure telephone signals, themethod comprising:receiving a predetermined tone signal; anddisconnecting the digital voice codec and connecting the secureinterface unit in response to receiving the predetermined tone signal.28. The method as defined in claim 27, wherein the predetermined tonesignal is one of a 2100 Hz and a pseudo 1800 Hz tone signal.
 29. Themethod as defined in claim 27, wherein the digital voice codec isdisconnected and the secure interface unit is connected within apredetermined time period from receiving the predetermined tone signal.30. The method as defined in claim 29, wherein the predetermined timeperiod is 250 ms.
 31. The method as defined in claim 27, furthercomprising receiving a first control packet confirming that a secureprotocol has been established in an external STU-III terminal system.32. The method as defined in claim 31, further comprising transmitting asecond control packet confirming that a secure protocol has beenestablished in the STU-III terminal system.
 33. The method as defined inclaim 32, wherein the second control packet is transmitted within apredetermined time period from receiving the first control packet. 34.The method as defined in claim 33, further comprising again receivingthe predetermined tone signal.
 35. The method as defined in claim 32,wherein the predetermined time period is 50 ms.
 36. The method asdefined in claim 31, further comprising receiving a secure protocolcontrol packet for initiating secure data transmission with the STU-IIIterminal system.
 37. The method as defined in claim 27, wherein thepredetermined tone signal is received via a satellite link.
 38. Themethod as defined in claim 27, further comprising transmitting a secureprotocol control packet indicating the capabilities of the STU-IIIterminal system.
 39. A communication system, comprising:a first securecommunication terminal for providing analog voiceband data; a firstprocessing circuit connected to receive the analog voiceband data fromthe first secure communication terminal and for converting the receivedanalog voiceband data into secure baseband data, said first processingcircuit transmitting the secure baseband data; a second processingcircuit connected to receive the transmitted secure baseband data fromthe first processing circuit, and for converting the received securebaseband data into analog voiceband data; a second secure communicationterminal for receiving the analog voiceband data from said secondprocessing circuit; and a satellite communication link; wherein saidsecure baseband data is transmitted from said first processing circuitvia said satellite communication link, and wherein at least one of saidfirst and second secure communication terminals is a STU-III terminal.40. The communication system as defined in claim 39, wherein said firstand second secure communication terminals are each a STU-III terminal.41. The communication system as defined in claim 39, wherein said secondprocessing circuit is disposed in a mobile earth station.
 42. Acommunication system, comprising:a first secure communication terminalfor providing interchangeably both secure and non-secure informationcarried by analog voiceband signal(s); a first processing circuitconnected to receive the analog voiceband signal(s) from the firstcommunication terminal and for converting the received analog voicebandsignal(s) into digital baseband data having a nominal rate of 9.6 kbit/sor less, said first, processing circuit operating interchangeably andautomatically on secure and non-secure information carried by analogvoiceband signals as a user of the first communication terminal chancesbetween secure and non-secure information, and said first processingcircuit providing the digital baseband data for transmission; a secondprocessing circuit connected to receive the transmitted digital basebanddata, and for converting the received digital baseband data into analogvoiceband signal(s), said second processing circuit operatinginterchangeably and automatically on the received secure and non-secureinformation corresponding to the secure and non-secure information,respectively, provided to the first processing circuit; and a secondsecure communication terminal for receiving the analog voicebandsignal(s) from said second processing circuit; wherein said firstprocessing circuit includes a signal discriminator for distinguishingbetween secure and non-secure information carried by analog voicebandsignal(s) received from said first secure communication terminal. 43.The communication system of claim 42, wherein said digital baseband datahas a nominal rate of 4.8 kbit/s.
 44. The communication system of claim42, wherein said digital baseband data has a nominal rate of 2.4 kbit/s.45. A communication system, comprising:a first secure communicationterminal for providing interchangeably both secure and non-secureinformation carried by analog voiceband signal(s); a first processingcircuit connected to receive the analog voiceband signal(s) from thefirst secure communication terminal and for converting the receivedanalog voiceband signal(s) into digital baseband data, said firstprocessing circuit operating interchangeably and automatically on secureand non-secure information carried by analog voiceband signals as a userof the first communication terminal changes between secure andnon-secure information, and said first processing circuit providing thedigital baseband data for transmission; a satellite communication link;a second processing circuit connected to receive the transmitted digitalbaseband data, and for converting the received digital baseband datainto analog voiceband signal(s), said second processing circuitoperating interchanqeably and automatically on the received secure andnon-secure information corresponding to the secure and non-secureinformation, respectively, provided to the first processing circuit; anda second secure communication terminal for receiving the analogvoiceband signal(s) from said second processing circuit; wherein saidfirst processing circuit includes a signal discriminator fordistinguishing between secure and non-secure information carried byanalog voiceband signal(s) received from said first secure communicationterminal, and wherein said digital baseband data is provided from saidfirst processing circuit to said second processing circuit via saidsatellite communication link.
 46. The communication system of claim 45,wherein said digital baseband data has a nominal rate of 9.6 kbit/s orless.
 47. The communication system of claim 46, wherein said digitalbaseband data has a nominal rate of 4.8 kbit/s.
 48. The communicationsystem of claim 46, wherein said digital baseband data has a nominalrate of 2.4 kbit/s.
 49. A communication unit comprising:a securecommunication terminal operable to perform at least one of (1) providingsecure information carried by analog voiceband signal(s); (2) receivingsecure information carried by analog voiceband signals(s); (3) providingnon-secure information carried by analog voicebind signals(s); and (4)receiving non-secure information carried by analog voiceband signal(s);and a processing circuit operable to perform interchangeably andautomatically a first, second, third and fourth function, wherein saidfirst function comprises: receiving secure information carried by analogvoiceband signal(s) output by the secure communication terminal,converting the received voiceband signal(s) carrying the secureinformation into digital baseband data having a nominal rate of 9.6kbit/s or less, and providing the digital baseband data fortransmission, wherein said second function comprises: receivingtransmitted secure information carried by digital baseband data having anominal rate of 9.6 kbit/s or less and converting the received digitalbaseband data into analog voiceband signal(s) carrying secureinformation to be output to the secure communication terminal, whereinsaid third function comprises receiving transmitted digital basebanddata containing non-secure information and having a nominal rate of 9.6kbit/s or less and converting the received digital baseband data intoanalog voiceband signal(s) carrying non-secure information to be outputto the secure communication terminal; and wherein said fourth functioncomprises receiving non-secure analog voiceband signal(s) carryingnon-secure information output by the secure communication terminal andconverting the received voiceband signal(s) carrying non-secureinformation into digital baseband data having a nominal rate of 9.6kbit/s or less, and providing the digital baseband data fortransmission.
 50. The communication unit of claim 49, wherein saidsecure and non-secure digital baseband data has a nominal rate of 4.8kbit/s.
 51. The communication unit of claim 49, wherein said secure andnon-secure digital baseband data has a nominal rate of 2.4 kbit/s. 52.The communication unit of claim 49, further comprising a satellitecommunication link, wherein the secure and non-secure digital basebanddata is transmitted via said satellite communication link.
 53. Thecommunication unit of claim 52, wherein said secure and non-securedigital baseband data has a nominal rate of 4.8 kbit/s.
 54. Thecommunication unit of claim 52, wherein said secure and non-securedigital baseband data has a nominal rate of 2.4 kbit/s.
 55. Thecommunication unit of claim 49, wherein said processing circuit furthercomprises a signal discriminator for distinguishing between secure andnon-secure analog voiceband signal(s) received from said securecommunication terminal.
 56. The communication unit of claim 55, whereinsaid secure and non-secure digital baseband data has a nominal rate of4.8 kbit/s.
 57. The communication unit of claim 55, wherein said secureand non-secure digital baseband data has a nominal rate of 2.4 kbit/s.58. The communication unit of claim 49, further comprising a satellitecommunication link, and wherein said processing circuit comprises asignal discriminator for distinguishing between secure and non-secureanalog voiceband signal(s) received from said secure communicationterminal, wherein the secure and non-secure digital baseband data istransmitted via said satellite communication link.
 59. The communicationunit of claim 58, wherein said secure and non-secure digital basebanddata has a nominal rate of 4.8 kbit/s.
 60. The communication unit ofclaim 58, wherein said secure and non-secure digital baseband data has anominal rate of 2.4 kbit/s.
 61. A communication method from a STU-IIIterminal over narrowband digital circuits, comprising:providing, fromsaid STU-III terminal, secure and non-secure information carried byanalog voiceband signals; distinguishing between the provided secure andnon-secure information carried by analog voiceband signals automaticallyand interchangeably; receiving the analog voiceband signal(s) andconverting the received analog voiceband signal(s) into digital basebanddata; and transmitting the digital baseband data.
 62. The method ofclaim 61, wherein said digital baseband data is transmitted at a nominalrate of 9.6 kbit/s or less.
 63. The method of claim 62, wherein saidtransmitted digital baseband data has a nominal rate of 4.8 kbit/s. 64.The method of claim 62, wherein said transmitted digital baseband datahas a nominal rate of 2.4 kbit/s.
 65. The method of claim 61, whereinsaid transmitting operation is performed via a communication satellitelink.
 66. The method of claim 65, wherein said digital baseband data istransmitted over said satellite link at a nominal rate of 9.6 kbit/s orless.
 67. The method of claim 66, wherein said digital baseband data hasa nominal rate of 4.8 kbit/s.
 68. The method of claim 66, wherein saiddigital baseband data has a nominal rate of 2.4 kbit/s.